Some audio digital signal processing (DSP) algorithms require audio signals to be delayed by less than an audio sample period. This is known as sub-sample delay. In a multichannel audio system, these algorithms may require different sub-sample delays on a per channel basis. Typically, this is accomplished within the DSP calculations, by passing the signals through special finite impulse response (FIR) filters. The resulting multi-channel output data (showing different sub-sample delays between its channels) is then sent to a set of digital-to-analog converters (DACs) that all run in a synchronized fashion, driven by identical master clocks and sample clocks. The resulting analog signals are then fed to drive a loudspeaker system. This approach has the disadvantage that the FIR filter introduces unwanted side effects into the signal, namely, ripple. A large (many taps) FIR filter will reduce the side effects but will require significant DSP resources, and so this forces trade-offs to be made between audio signal quality and DSP resources.